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Topic: What sample rate are you using? (Read 6701 times)

The only thing that's important is that you're happy with your recordings. I use 24/48, and I'm happy with that. Depending on how you record, and your editing process, would make a difference as to what you should be recording at. My suggestion, try out different things and determine what works best for you.

I record at 24/48, edit in floating point (some plugins doing their own upsampling), then use triangular dither to truncate back to 24bit files for output.. or SRC to 44.1 + triangular dither to truncate to 16 bit.

I use 24/48 for everything. Most of my stuff I upload to the LMA. I used to put up a 24/48 version and a 16/44 version but stopped doing that a while ago. I've found that most people just listen to the streaming mp3 on there anyways. And if archive my 24/48 raw files and the 24/48 FLAC. So if someone wants something different, I can convert the FLACs to whatever they need. I just had an artist ask me for mp3s of his solo show after I sent him the FLACs, so it took about 5 minutes to convert and zip them and send them to him.

I record at 24/44.1 if the eventual requirement is audio-only, and 24/48 if my sound track will be matched to video. Then I deliver whatever is specifically requested, whether it be mp3 files, CD-Rs, etc.

--aaronji wrote:

> A major advantage, as I understand it, is that the extra bits greatly reduce the noise introduced from quantization error during analog to digital conversion (each extra bit approximately halves the noise). So even if the recording peaks considerably below full-scale, you can amplify it without making the quantization noise audible...

Aaronji, that just isn't true as a rule. It all depends on the noise in the signal that's being quantized versus the converter's own noise floor. The quantization noise of the converter only needs to be a few dB lower than the noise in the incoming analog signal, however many bits that takes. Any further bits beyond that will (or should!) be random; therefore they don't contribute to the precision or accuracy of the recording in any way. The notion that computation (post-processing) will somehow work out more accurately with more bits is a total fallacy when those bits contain only noise.

Of course if they contain meaningful signal information, then by all means keep them--but from what I've read on this board over the years, I think that a lot of people here are kind of fooling themselves on that point. For live classical recording, which is what I mostly do, the signal going into the a/d converter might have a noise floor maybe 60 or (if I'm very lucky) 65 dB below the peak levels. 14-bit quantization can handle that (keeping in mind that the converter noise needs to be below the incoming noise at all frequencies). 16-bit recording allows for more conservative level settings and fewer accidental overloads--plus it's been decades since 14-bit recording was even an option on available recorders. But for example the BBC used 10-bit (analog companded) studio-to-transmitter links for their classical broadcasts for years, and they were justly renowned for their audio quality.

24-bit recording (which might have 18 or 19 actual bits to offer, when the noise level of the converters and the rest of the signal chain is taken into account) is even nicer than 16, but it's clearly a luxury for our benefit as live engineers. I use it because it doesn't hurt anything*, and once in a blue moon it helps a little. But I've never delivered a 24-bit recording to a client; people not directly involved in recording have no use for the extra data, which only causes them trouble and expense.

(Notes: If the extra, unneeded converter bits beyond the incoming signal's noise floor aren't random, then they're adding distortion to the recording, and any such converter should be removed from service immediately. And please keep in mind that the noise floor of a converter is whatever its ACTUAL value is, with the number of bits setting a theoretical limit that is never reached in practice; no physically realizable 24-bit converter has an analog input noise floor anywhere near 144 dB below peak level. So please don't make a mental leap from "24 bits" to "8 bits quieter than a 16-bit converter" because it's much more likely in reality to be only 2 or 3 bits quieter, if that.)

--best regards

_________* Edited later to add an asterisk above, and an important qualifying remark as a footnote: 24-bit recording "doesn't hurt anything" ONLY if you set your recording levels appropriately. At the risk of seeming to be unkind, let me say this as clearly as I can: If you're one of those people who completely misunderstand the concept of headroom, who actually AIM to achieve peak levels of -12 dB or even lower (instead of aiming to get your peak levels as close as possible to 0 dB without actually hitting it), then the use of 24-bit recording is hurting you by "enabling" your basic failure to understand the concept of recording levels, and your "24-bit" recordings are quite possibly noisier than they would be if you learned how digital recording actually works, and used 16-bit recording appropriately.

24-bit recording (which might have 18 or 19 actual bits to offer, when the noise level of the converters and the rest of the signal chain is taken into account) is even nicer than 16, but it's clearly a luxury for our benefit as live engineers. I use it because it doesn't hurt anything*, and once in a blue moon it helps a little.

I'll take it!

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But I've never delivered a 24-bit recording to a client; people not directly involved in recording have no use for the extra data, which only causes them trouble and expense.

D'oh!

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...So please don't make a mental leap from "24 bits" to "8 bits quieter than a 16-bit converter" because it's much more likely in reality to be only 2 or 3 bits quieter, if that.)

--best regards

_________* Edited later to add an asterisk above, and an important qualifying remark as a footnote: 24-bit recording "doesn't hurt anything" ONLY if you set your recording levels appropriately. At the risk of seeming to be unkind, let me say this as clearly as I can: If you're one of those people who completely misunderstand the concept of headroom, who actually AIM to achieve peak levels of -12 dB or even lower (instead of aiming to get your peak levels as close as possible to 0 dB without actually hitting it), then the use of 24-bit recording is hurting you by "enabling" your basic failure to understand the concept of recording levels, and your "24-bit" recordings are quite possibly noisier than they would be if you learned how digital recording actually works, and used 16-bit recording appropriately.

Thanks for another informative post DSatz! Just for max clarity, could you please elaborate on the * part?What would be the best practice for making use of 24-bit recording? Get as close as possible to 0 dBFS without going over, and don't just aim for a conservative -12 dB peak?I must admit, I'm not scared of the occasional red light flashing on a transient peak, and I'm happy to find out if I can hear my limiters in action. (mostly never!)

What would be the best practice for making use of 24-bit recording? Get as close as possible to 0 dBFS without going over, and don't just aim for a conservative -12 dB peak?I must admit, I'm not scared of the occasional red light flashing on a transient peak, and I'm happy to find out if I can hear my limiters in action. (mostly never!)

Get as close as possible to 0 dBFS without going over^This is the optimal strategy for all digital recording.. if only it were so simple in the real world, where we don't know in advance how loud it will get.

The common misconception DSatz is referring to is the notion that peaking no higher than -12dB or whatever is somehow beneficial in and of itself. Its not, its only good because it's not over. The goal isn't peaking specifically at -12dBfs, -6, -18, -3 or whatever number, the goal is simply not going over 0dBfs. The second (unstated) goal which this achieves at the same time is to try and keep levels high enough that the quietest parts do not drop beneath the recording system's noise-floor. That second part happens automatically if we push levels to get at close as possible to 0 dBfs, leaving almost all the "extra space" at the bottom as randomized noise. A portion of that extra stuff at the bottom gets tossed out when truncating the 24 bit file down to 16 bits, but not all of it. It's like trimming excess fat off that side, but still leaving a sufficeint fatty edge without cutting into the meaty portion. The full dynamics of the recording fits entirely within the 16bit file container. It just has less excess fat than before.

Find a comfortable balance between not going over at the top of the dynamic range nor dropping down beneath the noise floor of the recording system at the bottom of the range.^This strategy is effectively the same with respect to the end result, as long as the second half of the statement is true. And it's a lot easier to manage in the real world.

It leaves more space at the top and somewhat less at the bottom as the recording is being made. We can then trim some of that extra stuff from both the top and the bottom. We cut excess fat from both sides this time, and end up with the same full dynamic recording in the 16 bit file.

If you never have a problem with noise-floor in the quiet parts of your recordings, then there is no problem with peaking somewhat lower and amplifying later. It makes things easier in a practical sense by diminishing the possibility of overs.. or of limiters audibly cutting in. Clipping doesn't sound good, and neither do limiters if you can audibly hear them working. But otherwise there is no advantage to peaking lower other than not having to worry about overs.

No need to bump up too close to either end of the range. 17 or 18 bits worth of real-world recorded dynamic range leaves plenty of room in which to fit the full range of recorded sound with "headroom" and "footroom" to spare at both the top and bottom.

Sufficient headroom and footroom are not ends in themselves. It's fitting comfortably within the limits which is the real goal.

Hmm. I'm not sure what needs to be explained, because I don't know what you currently know and/or believe. How about this as an outline:

Any audio channel--whether you're using it for recording or just to pass a signal along from point "A" to point "B"--has some overload limit and some noise "floor." The difference or distance between those two levels is that channel's dynamic range.

Meanwhile, any audio signal that comes from acoustic reality will also have some maximum (peak) level and some noise floor of its own; thus it has a dynamic range as well.

We want to record (or transmit, or pass along) our audio signal such that its full, audible dynamic range is preserved.

As long as the channel's dynamic range exceeds the signal's dynamic range by at least a handful of dB--at all frequencies and at all times throughout the duration of the program--then that channel can preserve that signal's dynamic range without audible degradation. You'll need to set the level relationship between the signal and the channel pretty much as Gutbucket describes (don't let your overs go over, or your unders go under)--and then the signal coming out of the channel will retain, for all audible purposes, the dynamic range that it had on its way in.

Thus as long as the channel has sufficient dynamic range and the levels are set optimally--where the peaks of the program material come as close as possible to the channel's maximum level without actually reaching it--any additional extension of the channel's dynamic range can't audibly improve things, because the signal itself is the limiting factor at that point, not the channel.

This is actually nothing peculiar to digital audio; analog works exactly the same way. In fact I would strongly recommend "thinking in analog" in order to get the correct mental picture.

Maybe I misunderstood the initial point you were making. After reading the more recent posts, I'm assuming the initial point was that we shouldn't be using -12 (or -6 or -16, etc) dogmatically as our goal for where levels peak, but instead use 0. But, based on Gutbucket's comment and your last post, if we don't get the levels right at 0 and fall a little under, it's okay as long as the low parts of the audio signal still ride above the noise floor of the recorder. Is that (somewhat) accurate?

Heathen, that's exactly right as far as the general relationship between a signal and a channel is concerned.

However, a recorder's circuit design also matters. When the record level control on a recorder is set way too low, there's a real risk that unnecessary, extra noise is being added to the recording that doesn't come only from the noise floor of the a/d converter stage, but also from the noise floor of one or more earlier stages of the analog electronics. The user is basically inviting extra, unnecessary noise into the recording at each analog stage after the record level control.

One well-known recorder that I'm almost certain has this characteristic, for example, is the Sony PCM-M10. According to what I saw when I did some testing and measuring about five years ago, a 16-bit recording that reaches -2 on peaks would be considerably quieter than an exactly equivalent 24-bit recording that reaches only -12 but is then boosted 10 dB in post-processing. That's the opposite of the result that I think some people would expect.

I should probably repeat those measurements to make sure, but this isn't so much about that particular recorder as it is about the fact that there are always multiple analog stages before the a/d converters in any recorder--and if you set the record levels too low (or the sensitivity switch to its less sensitive setting when the more sensitive setting could be used safely without overload), it is rather likely that there will be unnecessary, audible noise contributions from one or more stages of the electronics prior to the converter.

Ideally, for the best signal-to-noise ratio. you want to nearly "saturate" each successive analog stage before the converters--keep them pumped full of signal, so that their noise doesn't make an audible contribution. Again, that's a basic way of thinking that an analog engineer would have from experience, that people have lost sight of in the digital era. I think some people might think about individual sample values too much, and/or think about them as if they only contain energy from the desired signal. But any one quantized sample reflects the instantaneous sum of the desired signal plus noise--and for any given sample, you can never know its "value due to signal" or its "value due to noise," just as you can't "unstir" a cup of coffee that already has cream and sugar in it.